Re: Mixing manually vs. Mixer AudioUnit and avoiding clipping
Re: Mixing manually vs. Mixer AudioUnit and avoiding clipping
- Subject: Re: Mixing manually vs. Mixer AudioUnit and avoiding clipping
- From: David Reaves <email@hidden>
- Date: Fri, 18 Jun 2010 22:07:41 +0200
When summing multiple sources of audio of equal volume, but whose
signals are NOT coherent (i.e., totally unrelated program in each
source), they will not sum to a 6dB (2.0x) increase as would identical
programs, but rather 3 dB (1.414x). In other words it is more like a
summation of power, rather than voltage.
So a mixer can sum channels, pretty reliably without clipping, with a
reduction of input gain of -3 dB (0.707) per input more than one (-3dB
* N-1 inputs).
Example:
2 sources summed: each source's gain in to the mixer should be dropped
by 3 dB (multiply each by 0.707).
3 sources: each should be dropped by 6 dB (multiply each by 0.707
squared, which is 0.5).
4 sources: each should be dropped by 9 dB (multiply each by 0.707
cubed, which is about 0.35), etc.
If you have, say, 30 inputs, you could seriously compromise the signal
to noise ratio by following that rule, so typically you use the number
of inputs that are *actually* used at any one time, rather than the
total number of available inputs.
Also, if you have adequate internal overhead above 1.0, you can more
simply multiply the summation rather than each individual input.
Dynamic processing is not needed or warranted.
David Reaves
On Fri, 18 Jun 2010 11:58:57 -0700, <email@hidden> wrote:
On Jun 18, 2010, at 11:44 AM, Gregory Wieber wrote:
You have to think in terms of the max output signal being "1". You
need to divide that by the number of busses you have and set the
volume for each bus accordingly.
I tried something like that but it didn't work very well. Take, for
instance, the case where you have one note playing, 5 notes are
silent and two notes are playing at very low levels. Dividing the
mixed signal by 8 is not going to sound right.
I think the trick is to reduce the volume of the very high notes,
but not to reduce the lower notes.
Google has pointed me to some articles on compression or Dynamic
Range Compression... maybe that's the direction I need to move in...
_______________________________________________
Do not post admin requests to the list. They will be ignored.
Coreaudio-api mailing list (email@hidden)
Help/Unsubscribe/Update your Subscription:
This email sent to email@hidden